1. Technical Field
The present invention relates to echo cancelling devices for use in signal transmission installations for full duplex transmission. It finds a particularly important application in telephony for solving problems raised by teleconference installations and so-called "free hand" telephone sets or at least having amplified listening. These problems are formed by danger of Larsen effect and the existence of an echo of acoustic origin.
The origin of such phenomena appears in FIG. 1 which shows, with continuous lines, the general diagram of a teleconference terminal. The signals x entering through a reception line LR and coming from a remote terminal 8 are amplified by an amplifier 10 and then broadcast in a listening hall by a loud-speaker 12. In the listening hall, the sound waves coming from the speaker whose speech is to be transmitted and from the loud-speaker 12 through acoustic coupling (shown schematically by a channel 13) are picked up by one or more microphones 14. Microphone 14 is connected to an amplifier 16 and the output signal y thereof is transmitted over the send line LE to the remote terminal. A listener-speaker placed at the remote terminal 8 will consequently hear not only the speech which is intended for him, but also an echo of his own speech with a delay proportional to the length of the lines LE and LR, perturbated by the transfer function of the acoustic channel 13. This echo is all the more troublesome the higher its level and the longer the delay. In satellite transmission, the delay may reach 600 ms and considerably disturb intelligibility. The Larsen effect occurs when the coupling formed by channel 13 is sufficiently tight for the gain in the loop formed of the two terminals and the lines exceeds 1.
2. Prior Art
Different devices have already been used for dealing with the acoustic echo and the Larsen effect, particularly automatic amplification gain controllers or echo cancelling devices in each terminal or "free hands" set.
The gain varying devices operate by introducing attenuation before the loud-speaker or after the microphone, depending on the detected communication direction. These devices have the drawback of producing a subjective impression of speech cut-off when the degree of inserted attenuation is high. Now, a high attenuation is necessary when the communication installations in which the delay is large and in which the echo must be greatly attenuated so as to remain tolerable.
The invention relates to echo cancelling devices for directly dealing with the echo by compensating for it by means of an equivalent signal and of opposite direction. For that, an echo cancelling device comprises an adaptive filtering device connected to the incoming signal input line for delivering an estimation of the echo of the signal received at the subtractive input of the subtractor which receives, at its additive input, the useful signal affected by the echo.
The principle of such a device is shown with broken lines in FIG. 1. The echo canceller comprises an adaptive filtering device FA which receives the incoming signal x and whose output feeds the subtractive input of a subtractor 18 whose additive input is connected to the output of the amplifier 16. The coefficients of filter FA are adapted automatically responsive to the signal e delivered to the line, equal to the difference between the echo-affected signal y of amplifier 16 and the estimated echo. The algorithm for adaptation of the coefficients of filter FA must be such that the filter models the characteristics of the acoustic coupling channel 13, formed by the beginning of the impulse response of this channel.
In the absence of speech from a speaker placed in the hall, the outgoing signal delivered to line LE is then reduced to an echo residue. When a speaker speaks in the hall in front of microphone 14, his speech is transmitted to a line LE without being attenuated or modified. Often, a device (not shown) is provided for detecting the presence of the speech signal coming from a speaker (for example by level detection) and for then momentarily blocking adaptation of filter FA so as to avoid any disturbance of the filter by the speech of the local speaker.
This solution has the great advantage, over automatic gain control of making bidirectional operation possible without attenuation of the useful signals, but presently existing devices which apply it are not entirely satisfactory, for their construction comes up against two main difficulties.
In the case of telephone installations, the acoustic echo cancellers comprise transversal adaptive digital filters operating at a sampling frequency from 8 to 16 kHz. Now, the impulse response of channel 13 is often very long and it corresponds to several thousands of coefficients at this sampling frequency. The computations to be carried out at each sampling time (filtering by convolution and adaptation of the coefficients of the filter) require a huge volume of operations. For carrying them very rapidly, numerous and expensive electronic circuits are indispensable.
The characteristics of the acoustic coupling channel 13 are time-variable, for example when a person moves in the listening hall. Theoretical reasons, related to the spectrum of the signal received over line LR and to the required length of the adaptive filter, limit the ability to track these variations (which must be taken immediately into account to avoid the untimely reappearance of the echo) and the initial convergence rate.
To overcome these problems an echo cancelling device has been proposed which is inserted between the line receiving the incoming signal and the line transmitting the outcoming signal so as to cancel out the echo, of the type comprising a plurality of processing channels assigned to successive adjacent sub-bands of the spectral band of the outgoing signal, each channel having:
a first analysis band-pass filter receiving the echo-affected signal to be transmitted, whose output is connected to the additive input of a subtractor;
a second analysis band-pass filter, identical to the first filter, receiving the incoming signal and feeding an adaptive filter delivering an estimated echo value in the sub-band to the subtractive input of the subtractor; and
a synthesis filter, symmetrical with the analysis filters and whose output feeds the transmission line.
Such a device is described for example in the article "Kompensation akustischer Echos in Frequenzteilbanden", Walter Kellermann, Frequenz, Vol. 39 (1985) No. 7-8, pp. 209-215. The general diagram of such a device is shown in FIG. 2. The signal x from line LR is fractionated by a bank of M analysis filters BFAR into M frequency sub-bands, usually all having the same width. The amplified signal y from the microphone is fractionated into M sub-bands 1, . . . , k, . . . , M by a bank of analysis filters BFAM identical to bank BFAR. In each sub-band of serial number k, an adaptive filter FA.sub.k is fed by the incoming signal x.sub.k and its output is subtracted by subtractor S.sub.k from the signal y.sub.k delivered by the analysis bank BFAM. As in the conventional canceller of FIG. 1, the adaptive filter FA.sub.k is adjusted so as to minimize the power of the signal e.sub.k at the output of subtractor S.sub.k.
Adaptation of the coefficients, shown schematically on FIG. 2 by an oblique arrow, is provided by a specific circuit using a conventional algorithm which will generally be the gradient algorithm although, in some cases, a simpler algorithm may be adapted, for example the sign algorithm, or another adaptation algorithm, for example the least squares algorithm.
The M signals e.sub.1, . . . , e.sub.k, . . . , e.sub.M feed a bank of synthesis filters BFS which rebuilds the full band signal sent as outgoing signal to the transmission line LE.
The device has several advantages compared with a conventional echo cancelling device, whose band is not fractionated:
The volume of computations to be effected per unit of time is considerably reduced since the signals in each sub-band can be sub-sampled: if Fe is the sampling frequency thought necessary for the full band signal, each of the M sub-bands of spectral width Fe/2M may theoretically be sub-sampled at its critical decimation frequency Fe/M. The volume of computations for the same impulse response time is theoretically divided by M, for the computation load in the analysis and synthesis filter banks is negligible as compared with the computations to be carried out in the adaptive filters.
In each sub-band k, the adaptation gain of the algorithm used may be optimized as a function of the power of signal x.sub.k in the band, which increases the convergence rate and the ability to track variations in the acoustic channel.
But these theoretical gains cannot be completely obtained in practice. A complete study of such a device A. Gilloire, Experiments with sub-band acoustic echo cancellers for teleconferencing, Proc. ICASSP-87, April 1987, Dallas, pp. 2141-2144) has shown that in fact it is not possible to adopt the critical decimation frequency if the sub-bands are directly adjacent. In fact, sub-sampling at the critical decimation frequency, required for reducing the computation volume as much as possible, cannot be used without causing spectrum aliasing to appear, at the frontiers between sub-bands, which are not cancelled by the adaptive filters. That leads either to using mutually separate sub-bands (without overlap), but with the drawback of introducing gaps in the spectrum of the signal reconstituted at the output of the synthesis banks BSF (which adversely affects the quality of speech when the number of sub-bands is high), or to sub-sampling the sub-bands at a frequency higher than the critical decimation frequency so as to form guard bands avoiding aliasing, which increases the computation speed required in the adaptive filters.